Method and system for releasing a voice response unit from a protocol session

ABSTRACT

An approach for processing voice calls over a packet switched network as to efficiently utilize the functionalities of a Voice Response Unit (VRU). According to one embodiment, a call originator, acting as a User Agent Client in accordance with the Session Initiation Protocol (SIP), issues messages to establish a first call-leg with the VRU. The VRU performs digit collection to obtain information to authenticate the call originator and to authorize the voice call. Based upon the issued messages from the call originator, the VRU establishes a second call-leg with the call terminator. The VRU is released from the voice call after binding the call-legs to connect the call originator to the call terminator.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to call processing, and is moreparticularly related to establishing a voice call over a packet switchednetwork via a voice response unit.

2. Discussion of the Background

The popularity and convenience of the Internet has resulted in thereinvention of traditional telephony services. These services areoffered over a packet switched network with minimal or no cost to theusers. IP (Internet Protocol) telephony, thus, have found significantsuccess, particularly in the long distance market. In general, IPtelephony, which is also referred to as Voice-over-IP (VOIP), is theconversion of voice information into data packets that are transmittedover an IP network. Users also have turned to IP telephony as a matterof convenience in that both voice and data services are accessiblethrough a single piece of equipment, namely a personal computer. Thecontinual integration of voice and data services further fuels thisdemand for IP telephony applications.

With the growing acceptance of IP telephony among the millions ofconsumers, service providers are cognizant of the impact that theseusers have on network capacity (e.g., switch sizing, line capacity) aswell as network resources (e.g., peripheral voice processing devices). Avaluable network resource is the voice response unit (VRU), whichprovides announcement and interactive voice response functions. Thesefunctions have become essential for the expedient treatment of voicecalls, especially in call center applications and operator assistance.Because VRU ports are expensive, it is desirable to ensure efficient useof such ports.

FIG. 8 illustrates a conventional IP telephony system. In this system800, an end office 801 houses a switch 803 and a VRU 805; the switch 803communicates with the VRU over a release line trunk (RLT). Switch 803serves user 807 to a public switch telephone network (PSTN) 809. The VRU805 is not functionally integrated with the IP network 815. That is, theVRU 805 works primarily in conjunction with the switch 803 within thePSTN realm. Using plain old telephone service (POTS), a calling party807 can place a telephone call over PSTN 809 to a called party 811 or813.

The PSTN 809 is connected to an IP (Internet Protocol) network 815,thereby enabling communication among the voice stations 807, 811, and813, which are connected to the public switch telephone network 809, andthe personal computers 817 and 819, which are attached to the IP network815. Attention is now drawn to transmission of voice calls over the IPnetwork 815.

Four possible scenarios exist with the placement of a VOIP call: (1)phone-to-phone, (2) phone-to-PC, (3) PC-to-phone, and (4) PC-to-PC. Inthe first scenario of phone-to-phone call establishment, voice station807 is switched through PSTN 809 by switch 803 to a VOIP gateway (notshown), which forwards the call through the IP network 815. Thepacketized voice call is then routed through the IP network 815, exitingthe IP network 815 at an appropriate point to enter PSTN 809 andterminates at voice station 811. Under the second scenario, voicestation 807 places a call to personal computer (PC) 817 through switch803 to PSTN 809. This voice call is then switched by the PSTN 809 to aVOIP gateway (not shown), which forwards the voice call to PC 817 via IPnetwork 815. The third scenario involves PC 817 placing a call to voicestation 813, for example. Using a voice encoder, PC 817 introduces astream of voice packets into IP network 815 that are destined for a VOIPgateway (not shown). A VOIP gateway (not shown) converts the packetizedvoice information into a POTS electrical signal, which is circuitswitched to voice station 813. Lastly, in the fourth scenario, PC 817establishes a voice call with PC 819. In this case, packetized voicedata is transmitted from PC 817 via IP network 815 to PC 819, where thepacketized voice data is decoded.

As indicated above, a network resource that permits the efficientprocessing of voice calls is a VRU 805. FIG. 9 shows a conventional callpath that is established by switch 803 to VRU 805. RLT links 901 connectswitch 803 to VRU 805, consuming two ports of each of these networkcomponents 803 and 805. RLT links 901 enable the release of a call backto switch 803 from VRU 805. This releasing functionality allows the VRU805 to be dropped from the voice call without impacting the callcompletion between call originator 903 and call terminator 905.

For explanatory purposes, it is assumed that a VRU 805 is needed toassist with call processing from call originator 903 (i.e., callingparty) to call terminator 805 (i.e., called party). Call originator 903places a voice call to switch 803 using port 1. In turn, the switch 803switches the call out of port 3 to port 1 of VRU 805. Once this call isestablished with VRU 805, the VRU 805 prompts the call originator 903,for example, to collect digits regarding account codes or billinginformation in order to authorize and validate the call originator 903.After this process, the VRU 805 loops the voice call back to the switch803 via port 2 over RLT 901 into port 4 of switch 803. Switch 803 thenswitches the call out of port 2 to call terminator 905. The RLT links901 permits the VRU 805 to drop out of the call when the call iscompleted between call originator 903 and call terminator 905. Thisrelease mechanism occurs over the PSTN 809. Such a mechanism isimportant because it frees up-the VRU 805 to process other calls; inaddition the switch 803 frees up two of its ports. An equivalentfunctionality is desirable in an IP telephony system.

Based on the foregoing, there is a clear need for improved approachesfor call processing with respect to use of network resources.

There is also a need to increase the integration of voice services overa data network.

There is a further need to minimize the cost of network operation.

Based on the need to efficiently employ network resources, an approachfor optimizing the use of VRU in an IP telephony environment is highlydesirable.

SUMMARY OF THE INVENTION

According to one aspect of the invention, a method is provided forprocessing a voice call over a packet switched network between a calloriginator and a call terminator. The method comprises establishing afirst call-leg between the call originator and a voice response unit(VRU) using a menu router that provides call control services accordingto a signaling protocol. The method also includes establishing a secondcall-leg between the VRU and the call terminator based upon thesignaling protocol. The method further includes binding the firstcall-leg and the second call-leg to complete the voice call between thecall originator and the call terminator, and releasing the voice callfrom the VRU based upon the signaling protocol. Under this approach,network resources are efficiently utilized, resulting in reduction ofnetwork operation costs.

According to another aspect of the invention, a communication system forprocessing a voice call over a packet switched network comprises a calloriginator that is configured to initiate and to receive the voice callover the packet switched network. A menu router performs call controlservices relating to the voice call. A voice response unit (VRU)processes a call setup request from the call originator. A callterminator is configured to process the voice call. The call originator,the call terminator, menu router, and the VRU communicate using a commonprotocol. The call originator establishes a first call-leg with the VRUvia the menu router. The VRU establishes a second call-leg with the callterminator and drops from the voice call upon binding the first call-legand the second call-leg. The above arrangement advantageously providesgreater integration of voice services over a packet switched network.

In yet another aspect of the invention, a computer-readable mediumcarrying one or more sequences of one or more instructions forprocessing a voice call over a packet switched network between a calloriginator and a call terminator. The one or more sequences of one ormore instructions include instructions which, when executed by one ormore processors, cause the one or more processors to perform the step ofestablishing a first call-leg between the call originator and a voiceresponse unit (VRU) using a menu router that provides call controlservices according to a signaling protocol. Another step comprisesestablishing a second call-leg between the VRU and the call terminatorbased upon the signaling protocol. Another step includes binding thefirst call-leg and the second call-leg to complete the voice callbetween the call originator and the call terminator. Yet another stepincludes releasing the voice call from the VRU based upon the signalingprotocol. This approach advantageously permits increased networkoperation efficiency.

BRIEF DESCRIPTION OF THE DRAWINGS

A more complete appreciation of the invention and many of the attendantadvantages thereof will be readily obtained as the same becomes betterunderstood by reference to the following detailed description whenconsidered in connection with the accompanying drawings, wherein:

FIG. 1 is a block diagram of an IP telephony system, according to anembodiment of the present invention;

FIG. 2 is diagram of the IP telephony protocol architecture employed bythe system of FIG. 1;

FIG. 3 is a diagram of a Session Initiation Protocol (SIP) model that isused in the system of FIG. 1;

FIG. 4 is a diagram of an exemplary network of menu routers, accordingto one embodiment of the present invention;

FIG. 5 is a diagram illustrating the interaction between the calloriginator, call terminator, and the voice response unit (VRU) in thesystem of FIG. 1;

FIG. 6 is a flow diagram of the operation of releasing the VRU in thesystem of FIG. 1;

FIG. 7 is a diagram of a computer system that can perform the process ofFIG. 6, in accordance with an embodiment of the present invention;

FIG. 8 is a diagram of a conventional IP telephony system; and

FIG. 9 is a diagram of the release line trunk (RLT) mechanism that isutilized in a traditional PSTN.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

In the following description, for the purpose of explanation, specificdetails are set forth in order to provide a thorough understanding ofthe invention. However, it will be apparent that the invention may bepracticed without these specific details. In some instances, well-knownstructures and devices are depicted in block diagram form in order toavoid unnecessarily obscuring the invention

The present invention accomplishes the release of VRU ports uponcompletion of the VOIP call by utilizing a signaling protocol, such as aSession Initiation Protocol (SIP). A call originator establishes a firstcall-leg with the VRU, which performs digit collection to obtain, forexample, account or billing information from the call originator. Basedupon the collected information, the VRU can determine whether the calloriginator is authorized to place a call. Thereafter, the VRUestablishes a second call-leg with the call terminator and drops out ofthe voice call.

Although the present invention is discussed with respect to the SessionInitiation Protocol, it should be appreciated that one of ordinary skillin the art would recognize that the present invention has applicabilityto other equivalent communication protocols. Further, the discussionbelow focuses on a call scenario that involves a PC-to-PC callestablishment, it is understood that the present invention can bepracticed with other call scenarios (e.g., PC-to-phone and phone-to-PC).

FIG. 1 shows the architecture of a IP telephony system according to oneembodiment of the present invention. Although call originator 101 andcall terminator 103 are shown to be attached to a IP network 105, it isunderstood that the call originator 101 and call terminator 103 may bevoice stations off the PSTN 107 as well. In general, the call originator101 and the call terminator 103 may be any device that is capable ofprocessing voice calls; e.g., an analog telephone set, a digitaltelephone station, or a personal computer that is loaded with theappropriate software and accompanying hardware. Voice communication canbe established in the IP telephony system 100 among any of the devices,101, 103, 109, and 111. However, this particular embodiment is explainedonly with respect to the communication between call originator 101 andcall terminator 103 in conjunction with VRU 113.

As shown in FIG. 1, the end office 151 houses a switch 115, whichbridges calls from the PSTN 107 to an automatic call distributor (ACD)117 via, for example, a release line trunk. A VRU controller 119 isconnected to ACD 117 through one or more Switch to Computer ApplicationInterface (SCAI) links. These SCAI links provide communication betweenACD 117 and VRU controller 119, which is responsible for selecting agroup or a particular agent to which the call is to be routed. In otherwords, the VRU controller 119 communicates with the ACD 117 for calldelivery to the different agents within, for example, an operator center(not shown). The term agent denotes an entity that participates in callprocessing; e.g., a live person on a manual operator console or asoftware process. VRU controller 119 further provides suchfunctionalities as coordinating data and voice for operator-assistedcalls. A Local Area Network (LAN) 121 permits the VRU controller 119 tocommunicate with VRU 113. LAN 121 also provides connectivity to IPnetwork 105.

As previously mentioned, call originator 101 and call terminator 103 arePCs that have access to IP network 105. It is assumed that these devices101 and 103 are appropriately equipped with voice encoders and decodersas well as software to process VOIP calls. In this example, calloriginator 101 initiates a VOIP call that requires the services of a VRU113.

VRU 113 provides announcement capability as well as Interactive VoiceResponse (IVR) capability. In essence, VRU 113 provides an ability tocollect various information from and supply announcement information toa calling party (i.e., call originator). In this instance, after thecall originator 101 establishes a call-leg with VRU 113, the VRU 113prompts call originator 101 for a billing code or an account code,thereby enabling the authentication and validation of call originator101 to authorize the desired voice call. After being grantedauthorization, the VOIP call from call originator 101 can be completedto call terminator 103 through the IP network 105.

During the call process of the VOIP call from call originator 101 tocall terminator 103, it is important that VRU 113 be utilizedefficiently. Keeping the VRU 113 in the voice call for the entire callwould result in the VRU 113 remaining idle for a significant portion ofthat voice call, wasting precious network resources. That is, VRU 113should remain in the voice call only for the duration that it is neededto collect information from call originator 101. To accomplish thistask, a suite of protocols are utilized that collectively define IPtelephony signaling.

FIG. 2 illustrates the IP telephony protocol architecture in accordancewith an embodiment of the present invention. The layered nature of thearchitecture provides protocol separation and independence, whereby oneprotocol can be exchanged or modified without affecting the other higherlayer or lower layer protocols. It is advantageous that the developmentof these protocols can occur concurrently and independently.

The foundation of the architecture rests with the IP layer 201. The IPlayer 201 provides an unreliable, connectionless data delivery serviceat the network level. The service is “unreliable” in the sense that thedelivery is on a “best effort” basis; that is, no guarantees of packetdelivery are made. IP is the de facto Internet working protocolstandard. Current standards provide two versions of IP: Version 4 andVersion 6. One of the key differences between the versions concernsaddressing; under Version 4, the address fields are 32 bits in length,whereas in Version 6, the address field has been extended to 128 bits.

Above the IP layer 201 are the TCP (Transmission Control Protocol) 203and the UDP (User Datagram Protocol) 205. The TCP layer 203 provides aconnection-oriented protocol that ensures reliable delivery of the IPpackets, in part, by performing sequencing functions. This sequencingfunction reorders any IP packets that arrive out of sequence. Incontrast, the User Datagram Protocol (UDP) 205 provides a connectionlessservice that utilizes the IP protocol 201 to send a data unit, known asa datagram. Unlike TCP 203, UDP 205 does not provide sequencing ofpackets, relying on the higher layer protocols to sort the information.UDP 205 is preferable over TCP 203 when the data units are small, whichsaves processing time because of the minimal reassembly time. One ofordinary skill in the art would recognize that embodiments of thepresent invention can be practiced using either TCP 203 or UDP 205, aswell as other equivalent protocols.

The next layer in the IP telephony architecture of FIG. 2 supplies thenecessary IP telephony signaling and includes the H.323 protocol 207 andthe Session Initiation Protocol (SIP) 209. The H.323 protocol 207, whichis promulgated by the International Telecommunication Union (ITU),specifies a suite of protocols for multimedia communication. SIP 209 isa competing standard that has been developed by the Internet EngineeringTask Force (IETF). SIP 209 is a signaling protocol that is based on aclient-server model. It should be noted that both the H.323 protocol 207and SIP 209 are not limited to IP telephony applications, but haveapplicability to multimedia services in general. In the preferredembodiment of the present invention, SIP 209 is used to create andterminate voice calls over an IP network 105. However, it is understoodthat one of ordinary skill in the art would realize that the H.323protocol 207 and similar protocols can be utilized in lieu of SIP 209.Above SIP 209 is the Session Description Protocol (SDP) 211, whichprovides information about media streams in the multimedia sessions, asto permit the recipients of the session description to participate inthe session.

As seen in FIG. 2, SIP 209 can utilize either TCP 203 or UDP 205.However, UDP 205 is adopted in the preferred embodiment of the presentinvention. Similar to other IETF protocols (e.g., the simple mailtransfer protocol (SMTP) and Hypertext Transfer Protocol (HTTP)), SIP209 is a textual protocol. As indicated earlier, SIP 209 is aclient-server protocol, and as such, clients generate requests that areresponded to by the servers.

FIG. 3 illustrates the client-server model 300 of SIP 209. On the clientside, SIP 209 defines a User Agent Client (UAC), which is responsiblefor initiating a SIP request. On the server side, a User Agent Server(UAS) 303 receives the SIP request and returns an appropriate response.Both the UAC 301 and the UAS 303 act on behalf of an end user. SIPfurther defines two types of User Agent Servers: (1) proxy server 305,and (2) redirect server 307.

SIP proxy server 305 receives requests from the UAC 301 and determinesthe next server that the request should be forwarded. Accordingly, theSIP proxy server 305 sends the request to such a server. During thisprocess of receiving and forwarding, the proxy server 305 behaves bothas a client and a server by issuing both requests and responses asappropriate.

In the case of the redirect server 307, the client 301 is given greaterresponsibility. The redirect server 307 does not forward requests fromUAC 301 to the next server, but instead responds back to the client 301with the address of the next server. The client 301, thus, has the onusof directly communicating with this designated server. Using the SIPclient-server model, IP telephony calls can be processed, according tothe present invention, to efficiently utilize a Voice Response Unit.

FIG. 4 shows a computer network associated with the call originator inimplementing SIP, according to an embodiment of the present invention.Call originator 101, which in this exemplary embodiment is a PC, isattached to LAN 401. However, it is recognized that the call originator101 can be any device that is capable of supporting IP voice. LAN 401can be any type of network, including Ethernet, Token Ring, FDDI (FiberDistributed Data Interface), or ATM (Asynchronous Transfer Mode). Inthis exemplary network, the call originator (as a User Agent Client) 101communicates with a menu router/proxy server 405, which acts as a UAS.The menu router 405 offers call originator 101 a menu of choices thatinvoke various call processing actions. Additionally, menu router 405launches specific menu scripts according to the request message that issent by call originator 101. The menu router 405 provides media proxyand media mixing and can perform as a proxy server according to the menuscripts. Because the menu router 405 has the capability to behave as aproxy server, the menu router 405 is also designated as a menurouter/proxy server. Although the menu router 405 is shown as a part ofthe same network as the call originator 101, the menu router 405 mayexist anywhere within the same network domain as the call originator.

When UAC 403 issues a request, call originator 101 first locates a proxyserver 405 using the IP address of the proxy server 405. Assuming theVOIP call is destined for call terminator 103, proxy server 405 forwardsa request from call originator 101 to proxy server 407. To reach proxyserver 407, the request travels over LAN 401 to a gateway 409, whichprovides an interface to IP network 105. After traversing the IP network105, the request emerges at another gateway 403, which is attached toLAN 411, where the request is retrieved by proxy server 407. Proxyserver 407 then communicates with a location server 413 to determine thelocation of call terminator 103.

Attention is now drawn to a VOIP call involving VRU 113, as shown inFIG. 5. For explanatory purposes, it is assumed that the call originator101, the call terminator 103, and the VRU 113 belong to separate domains501, 503, and 505, respectively. Within domain 501, call originator 101sends a request to establish a call with VRU 113 to menu router/proxyserver 405, which in turn communicates with menu router/proxy server 507of domain 505. The proxy server 507 notifies VRU 113 of the request bycall originator 101. Upon receiving the request from call originator101, proxy server 507 inquires location server 509 for the address ofVRU 113. If VRU 113 is able to accept the request (i.e., has availableports), VRU 113 issues an acknowledgment back to call originator 101.Consequently, a successful connection has been made between calloriginator 101 and VRU 113 and thus, VRU 113 can begin the process ofdigit collection, as previously discussed.

Upon completion of the digit collection from call originator 101, VRU113 issues a request to proxy server 507 to establish a call-leg withcall terminator 103 within domain 503. After receiving the request fromthe VRU 113, proxy server 407 queries location server 511 to determinethe address of call terminator 103. Subsequently, call terminator 103receives the request and acknowledges, thereby establishing a call-legbetween call terminator 103 and VRU 113. Having established this secondcall-leg, VRU 113 binds the first call-leg from call originator 101 tothis second call-leg to permit the communication between call originator101 and call terminator 103. VRU 113 then drops from the call. Bydropping from the call, VRU 113 frees up its ports to process othervoice calls. It should be noted that within domain 501, there exists alocation server 513 to process calls for device 101; in actualimplementation, call originator 101 can also behave as a call terminatorwithin the single device.

The system of FIG. 5 employs SIP to exchange messages among domains 501,503, and 505. A detailed discussion of SIP and its call control servicesare described in IETF RFC 2543 and IETF Internet draft “SIP Call ControlServices”, Jun. 17, 1999; both of these documents are incorporatedherein by reference in their entirety. SIP messages are either requestsor responses. The User Agent Clients issue requests, while the UserAgent Servers provide responses to these requests. SIP defines six typesof requests, which are also referred to as methods. The first method isthe INVITE method, which invites a user to a conference. The next methodis the ACK method, which provides for reliable message exchanges forinvitations in that the client is sent a confirmation to the INVITErequest. That is, a successful SIP invitation includes an INVITE requestfollowed by an ACK request.

Another method is the BYE request, which indicates to the UAS that thecall should be released. In other words, BYE terminates a connectionbetween two users or parties in a conference. The next method is theOPTIONS method; this method solicits information about capabilities anddoes not assist with establishment of a call. Lastly, the REGISTERprovides information about a user's location to a SIP server.

FIG. 6 shows the operation involving the use of a VRU to establish acall between a call originator 101 and a call terminator 103, utilizingSIP. It should be noted that FIG. 6 provides a simplified SIP messageflow between call originator 101 and call originator 103 using VRU 113.In step 601, call originator 101 issues an INVITE request to VRU 113. Inresponse, VRU 113 issues a 200 OK message, indicating that theinvitation was successful, per step 603. Next, call originator 101, asin step 605, sends an ACK message to VRU 113 to acknowledge the previousmessage. At this point, call originator 101 and VRU 113 exchange data asnecessary.

After the VRU 113 completes processing, call originator 101 issues a BYEVRU message using an Also header to indicate that call originator 101seeks to establish a call with call terminator 103, per step 607. Next,VRU 113 issues a 200 OK message, as in step 609, to indicate that theprevious message was successful.

In turn, VRU 113, as in step 611, sends an INVITE call terminatormessage to call terminator 103. In step 613, call terminator 103 issuesa 200 OK message to VRU 113, which then acknowledges via an ACK message,per step 615. At this point in the call processing, VRU 113 drops out ofthe voice call, leaving call originator 101 and call terminator 103 toexchange voice messages. Under this arrangement, the valuable networkresource, VRU 113, is not unnecessarily tied up for the duration of thevoice call between call originator 101 and call terminator 103.

FIG. 7 illustrates a computer system 701 upon which an embodimentaccording to the present invention may be implemented. Computer system701 includes a bus 703 or other communication mechanism forcommunicating information, and a processor 705 coupled with bus 703 forprocessing the information. Computer system 701 also includes a mainmemory 707, such as a random access memory (RAM) or other dynamicstorage device, coupled to bus 703 for storing information andinstructions to be executed by processor 705. In addition, main memory707 may be used for storing temporary variables or other intermediateinformation during execution of instructions to be executed by processor705. Computer system 701 further includes a read only memory (ROM) 709or other static storage device coupled to bus 703 for storing staticinformation and instructions for processor 705. A storage device 711,such as a magnetic disk or optical disk, is provided and coupled to bus703 for storing information and instructions.

Computer system 701 may be coupled via bus 703 to a display 713, such asa cathode ray tube (CRT), for displaying information to a computer user.An input device 715, including alphanumeric and other keys, is coupledto bus 703 for communicating information and command selections toprocessor 705. Another type of user input device is cursor control 717,such as a mouse, a trackball, or cursor direction keys for communicatingdirection information and command selections to processor 705 and forcontrolling cursor movement on display 713.

Embodiments are related to the use of computer system 701 to control ARU201 remotely via the transmission of control messages. According to oneembodiment, the issuance of SIP messages is provided by computer system701 in response to processor 705 executing one or more sequences of oneor more instructions contained in main memory 707. Such instructions maybe read into main memory 707 from another computer-readable medium, suchas storage device 711. Execution of the sequences of instructionscontained in main memory 707 causes processor 705 to perform the processsteps described herein. One or more processors in a multi-processingarrangement may also be employed to execute the sequences ofinstructions contained in main memory 707. In alternative embodiments,hard-wired circuitry may be used in place of or in combination withsoftware instructions. Thus, embodiments are not limited to any specificcombination of hardware circuitry and software.

Further, the Sessions Initiation Protocol as well as the instructions totransmit and receive SIP messages may reside on a computer-readablemedium. The term “computer-readable medium” as used herein refers to anymedium that participates in providing instructions to processor 705 forexecution. Such a medium may take many forms, including but not limitedto, non-volatile media, volatile media, and transmission media.Non-volatile media includes, for example, optical or magnetic disks,such as storage device 711. Volatile media includes dynamic memory, suchas main memory 707. Transmission media includes coaxial cables, copperwire and fiber optics, including the wires that comprise bus 703.Transmission media can also take the form of acoustic or light waves,such as those generated during radio wave and infrared datacommunications.

Common forms of computer-readable media include, for example, a floppydisk, a flexible disk, hard disk, magnetic tape, or any other magneticmedium, a CD-ROM, any other optical medium, punch cards, paper tape, anyother physical medium with patterns of holes, a RAM, a PROM, and EPROM,a FLASH-EPROM, any other memory chip or cartridge, a carrier wave asdescribed hereinafter, or any other medium from which a computer canread.

Various forms of computer readable media may be involved in carrying oneor more sequences of one or more instructions to processor 705 forexecution. For example, the instructions may initially be carried on amagnetic disk of a remote computer. The remote computer can load theinstructions relating to the transmission of SIP messages to controlcall processing remotely into its dynamic memory and send theinstructions over a telephone line using a modem. A modem local tocomputer system 701 can receive the data on the telephone line and usean infrared transmitter to convert the data to an infrared signal. Aninfrared detector coupled to bus 703 can receive the data carried in theinfrared signal and place the data on bus 703. Bus 703 carries the datato main memory 707, from which processor 705 retrieves and executes theinstructions. The instructions received by main memory 707 mayoptionally be stored on storage device 711 either before or afterexecution by processor 705.

Computer system 701 also includes a communication interface 719 coupledto bus 703. Communication interface 719 provides a two-way datacommunication coupling to a network link 721 that is connected to alocal network 723. For example, communication interface 719 may be anetwork interface card to attach to any packet switched local areanetwork (LAN). As another example, communication interface 719 may be anasymmetrical digital subscriber line (ADSL) card, an integrated servicesdigital network (ISDN) card or a modem to provide a data communicationconnection to a corresponding type of telephone line. Wireless links mayalso be implemented. In any such implementation, communication interface719 sends and receives electrical, electromagnetic or optical signalsthat carry digital data streams representing various types ofinformation.

Network link 721 typically provides data communication through one ormore networks to other data devices. For example, network link 721 mayprovide a connection through local network 723 to a host computer 725 orto data equipment operated by a service provider, which provides datacommunication services through the IP network 105. LAN 723 and IPnetwork 105 both use electrical, electromagnetic or optical signals thatcarry digital data streams. The signals through the various networks andthe signals on network link 721 and through communication interface 719,which carry the digital data to and from computer system 701, areexemplary forms of carrier waves transporting the information. Computersystem 701 can send SIP messages and receive data, including programcode, through the network(s), network link 721 and communicationinterface 719.

The techniques described herein provide several advantages over priorapproaches to call processing in which a VRU 113 is needed to establisha VOIP call between call originator 101 and call terminator 103. Thepresent invention presents an efficient and economically feasibleapproach to processing VOIP calls involving a VRU 113. The VRU 113 dropsfrom the voice call after binding the two call-legs of the calloriginator 101 and the call terminator 103.

Obviously, numerous modifications and variations of the presentinvention are possible in light of the above teachings. It is thereforeto be understood that within the scope of the appended claims, theinvention may be practiced otherwise than as specifically describedherein.

What is claimed as new and desired to be secured by Letters Patent ofthe United States is:
 1. A method for processing a voice call over apacket switched network between a call originator and a call terminator,the method comprising: establishing a first call-leg between the calloriginator and a voice response unit (VRU) over the packet switchednetwork using a menu router that provides call control servicesaccording to a signaling protocol; establishing a second call-legbetween the VRU and the call terminator over the packet switched networkbased upon the signaling protocol; binding the first call-leg and thesecond call-leg to complete the voice call between the call originatorand the call terminator over the packet switched network; and releasingthe voice call from the VRU based upon the signaling protocol.
 2. Themethod according to claim 1, wherein the signaling protocol in the stepof establishing the first call-leg is a Session Initiation Protocol. 3.The method according to claim 1, further comprising: collecting digitinformation from the call originator by the VRU; and performingauthentication and authorization of the call originator by the VRU basedupon the collecting step.
 4. The method according to claim 1, whereinstep of establishing the first call-leg comprises: invoking menu scriptswithin the menu router based upon messages conveyed by the signalingprotocol; and providing media proxy and media mixing services.
 5. Themethod according to claim 1, wherein the packet switched network is anInternet Protocol (IP) network.
 6. The method according to claim 1,wherein the packet switched network is the Internet.
 7. The methodaccording to claim 1, wherein the call originator and the callterminator are personal computers with voice processing capabilities. 8.The method according to claim 7, wherein the step of establishing thefirst call-leg comprises locating the VRU by querying a location serverthat stores an address of the VRU.
 9. A communication system forprocessing a voice call over a packet switched network, thecommunication system comprising: a call originator configured toinitiate and to receive the voice call over the packet switched network;a menu router configured to perform call control services relating tothe voice call; a voice response unit (VRU) configured to process a callsetup request from the call originator; and a call terminator configuredto process the voice call, wherein the call originator, the callterminator, menu router, and the VRU communicate using a commonprotocol, the call originator establishing a first call-leg with the VRUover the packet switched network via the menu router, the VRUestablishing a second call-leg with the call terminator over the packetswitched network and dropping from the voice call upon binding the firstcall-leg and the second call-leg.
 10. The system according to claim 9,the common protocol is a Session Initiation Protocol.
 11. The systemaccording to claim 9, wherein the VRU is configured to collect digitinformation from the call originator and to perform authentication andauthorization of the call originator based upon the collectedinformation.
 12. The system according to claim 9, wherein the menurouter is configured to invoke menu scripts based upon messages that areconveyed by the common protocol, the menu router providing media proxyand media mixing services to the call originator and the callterminators.
 13. The system according to claim 9, wherein the packetswitched network is an Internet Protocol (IP) network.
 14. The systemaccording to claim 9, wherein the packet switched network is theInternet.
 15. The system according to claim 9, wherein the calloriginator and the call terminator are personal computers with voiceprocessing capabilities.
 16. The system according to claim 9, furthercomprising: a first location server configured to provide addressinginformation of the VRU, the first location server being in a domain thatincludes the VRU; a second location server configured to provideaddressing information of the call terminator, the second locationserver being in another domain that includes the call terminator. 17.The system according to claim 9, further comprising: a switch coupled tothe VRU and configured to switch the voice call over a voice network;and a voice station configured to initiate the voice call that isswitched by the switch onto the packet switched network.
 18. Acomputer-readable medium carrying one or more sequences of one or moreinstructions for processing a voice call over a packet switched networkbetween a call originator and a call terminator, the one or moresequences of one or more instructions including instructions which, whenexecuted by one or more processors, cause the one or more processors toperform the steps of: establishing a first call-leg between the calloriginator and a voice response unit (VRU) over the packet switchednetwork using a menu router that provides call control servicesaccording to a signaling protocol; establishing a second call-legbetween the VRU and the call terminator over the packet switched networkbased upon the signaling protocol; binding the first call-leg and thesecond call-leg to complete the voice call between the call originatorand the call terminator over the packet switched network; and releasingthe voice call from the VRU based upon the signaling protocol.
 19. Thecomputer-readable medium according to claim 18, wherein the signalingprotocol is a Session Initiation Protocol.
 20. The computer-readablemedium according to claim 18, further comprising performing the stepsof: collecting digit information from the call originator by the VRU;and performing authentication and authorization of the call originatorby the VRU based upon the collecting step.
 21. The computer-readablemedium according to claim 18, wherein step of establishing the firstcall-leg comprises: invoking menu scripts within the menu router basedupon messages conveyed by the signaling protocol; and providing mediaproxy and media mixing services.
 22. The computer-readable mediumaccording to claim 18, wherein the step of establishing the firstcall-leg comprises locating the VRU by querying a location server thatstores an address of the VRU.